Integrated Services Digital Network (ISDN) is a set of communications standards for

simultaneous digital transmission of voice, video, data, and other network services. The
key feature of ISDN is that it integrates speech and data on the same lines.

Integrated Services Digital Network (ISDN) is comprised of digital telephony

and data-transport services offered by regional telephone carriers. ISDN involves
the digitalization of the telephone network, which permits voice, data, text,
graphics, music, video, and other source material to be transmitted over existing
telephone. The emergence of ISDN represents an effort to standardize
subscriber services, user/network interfaces, and network and internetwork
capabilities. ISDN applications include high-speed image applications (such as
Group IV facsimile), additional telephone lines in homes to serve the
telecommuting industry, high-speed file transfer, and video conferencing. Voice
service is also an application for ISDN. This chapter summarizes the underlying
technologies and services associated with ISDN.

Difference Between ISDN and PSTN

1. PSTN lines аrе analogue whіƖе ISDN lines аrе digital.

2.Whеn comparing thе two networks, thе PSTN lines аrе used fοr small companies аnԁ
ISDL аrе used fοr bіɡɡеr companies.
3.Thе ISDN provides 128 kbit/s, whісh іѕ really ɡοοԁ fοr thе Internet. PSTN hаѕ a
disadvantage thаt іt ԁοеѕ nοt mаkе thе mοѕt possible υѕе οf thе broadband.
4.WhіƖе PSTN ԁοеѕ nοt allow two simultaneous connections, іt іѕ allowed іn ISDN
service.
5.Whеn using ISDN, one саn mаkе fаѕtеr calls thаn whеn using thе PSTN.

• Global Navigation Satellite Systems (GNSS) is the standard generic term for
satellite navigation systems (Sat Nav) that provide autonomous geo-spatial
positioning with global coverage. GNSS allows small electronic receivers to
determine their location (longitude, latitude, and altitude) to within a few metres
using time signals transmitted along a line-of-sight by radio from satellites.
Receivers calculate the precise time as well as position, which can be used as a
reference for scientific experiments. The Global Positioning System (GPS) is the
only fully functional Global Navigation Satellite System (GNSS). The GPS uses a
constellation of between 24 and 32 Medium Earth Orbit satellites that transmit
precise microwave signals, that enable GPS receivers to determine their location,
speed, direction, and time. GPS was developed by the United States Department
of Defense..

In radio communications, a base station is a wireless communications station installed at

a fixed location and used to communicate as part of either:

• a push-to-talk two-way radio system, or;

• a wireless telephone system such as cellular CDMA or GSM cell site.
 • A wireless telephone base station communicates with a mobile or hand-held
phone. For example, in a wireless telephone system, the signals from one or more
mobile telephones in an area are received at a nearby base station, which then
connects the call to the land-line network

• A cell site is a term used primarily in North America for a site where antennas
and electronic communications equipment are placed on a radio mast or tower to
create a cell in a cellular network. A cell site is composed of a tower or other
elevated structure for mounting antennas, and one or more sets of
transmitter/receivers transceivers, digital signal processors, control electronics, a
GPS receiver for timing (for CDMA2000 or IS-95 systems), regular and backup
electrical power sources, and sheltering. A synonym for "cell site" is "cell tower",
although many cell site antennas are mounted on buildings rather than as towers.
In GSM networks, the technically correct term is Base Transceiver Station (BTS).

Cellular network tower in the Philippines.

• [edit] Channel reuse- Cellular

networks are designed to create a mass
communication solution from a limited
amount of channels (slices of radio
frequency spectrum necessary to make one conversation) that are licensed to an
operator of a cellular service. To overcome this limitation, it is necessary to repeat
and reuse the same channels at different locations.

where R is the cell radius and N is the number of cells per cluster. Cells may vary in
radius in the ranges (1 km to 30 km). The boundaries of the cells can also overlap
between adjacent cells and large cells can be divided into smaller cells.

Handover - In a cellular system, as the distributed mobile transceivers move from cell to
cell during an ongoing continuous communication, switching from one cell frequency to
a different cell frequency is done electronically without interruption and without a base
station operator or manual switching. This is called the handover or handoff. Typically, a
new channel is automatically selected for the mobile unit on the new base station which
will serve it. The mobile unit then automatically switches from the current channel to the
new channel and communication continues.
Full Rate - Full Rate or FR or GSM-FR or GSM 06.10 was the first digital speech
coding standard used in the GSM digital mobile phone system. The bit rate of the codec
is 13.2 kbit/s, or 1.65 bits/audio sample.

The speech is analog, so in order to be transmitted over digital

communication it should be digitized. The method used by GSM is Regular
Pulse Excited – Linear Predictive Coder (RPE-LPC) with a Long Term
Predictor loop. The main idea behind this smart-looking name is simple.
Speech is divided into 20 millisecond samples; current sample may be
predicted from previous samples, that’s due to slow change of voice
patterns. Predicted and real information are compared and the difference is
saved. Each 20-millisecond sample is encoded using 260 bits (that requires
13 kbps). Testing let to distinguish three classes of bits out of 260, that are
classified by they sensitivity to errors. The most sensitive class has CRC
and together with moderate sensitivity class is encoded using ½ rate
convolutional encoder of length 4 – each input bit is encoded as two bits,
based on 4 previous bits. Thus we have 456 bits per 20 milliseconds
sample (that requires 22.8 kbps).

Discontinuous transmission

The idea is based on the fact that a person speaks less than 40% of time in
normal conversation, so turning the transmitter off can save power. In
order to distinguish voice and background noise, very accurate Voice
Activity Detector should be used. While transmitter is off, the receiving end
will hear a total silence, that’s due to digital transmission. To avoid this,
comfort noise is generated trying to match the characteristics of
background noise.

multiplexing (also known as muxing) is a process where multiple analog message

signals or digital data streams are combined into one signal over a shared medium. The
aim is to share an expensive resource. For example, in telecommunications, several
phone calls may be transferred using one wire. The multiplexed signal is transmitted over
a communication channel, which may be a physical transmission medium. The
multiplexing divides the capacity of the low-level communication channel into several
higher-level logical channels, one for each message signal or data stream to be
transferred. A reverse process, known as demultiplexing, can extract the original channels
on the receiver side. A device that performs the multiplexing is called a multiplexer
(MUX), and a device that performs the reverse process is called a demultiplexer
(DEMUX). Multiplexers are used to connect terminals located throughout a
building to a central mainframe. They are also used to connect terminals located
at remote locations to a central mainframe through the phone lines.
Types of multiplexing

space-division multiplexing (SDM)

frequency-division multiplexing (FDM)

Frequency division multiplexing (FDM) means that the total bandwidth available to the
system is divided into a series of nonoverlapping frequency sub-bands that are then
assigned to each communicating source and user pair. Figures 7-7a and 7-7b show how
this division is accomplished for a case of three sources at one end of a system that are
communicating with three separate users at the other end. Note that each transmitter
modulates its source's information into a signal that lies in a different frequency sub-band
(Transmitter 1 generates a signal in the frequency sub-band between 92.0 MHz and 92.2
MHz, Transmitter 2 generates a signal in the sub-band between 92.2 MHz and 92.4 MHz,
and Transmitter 3 generates a signal in the sub-band between 92.4 MHz and 92.6 MHz).
The signals are then transmitted across a common channel.

[+] Enlarge Image

Figure 7-7a—A system using frequency division multiplexing.

 Figure 7-7b—Spectral occupancy of signals in an FDM system.

At the receiving end of the system, bandpass filters are used to pass the desired signal
(the signal lying in the appropriate frequency sub-band) to the appropriate user and to
block all the unwanted signals. To ensure that the transmitted signals do not stray outside
their assigned sub-bands, it is also common to place appropriate passband filters at the
output stage of each transmitter. It is also appropriate to design an FDM system so that
the bandwidth allocated to each sub-band is slightly larger than the bandwidth needed by
each source. This extra bandwidth, called a guardband, allows systems to use less
expensive filters (i.e., filters with fewer poles and therefore less steep rolloffs).

FDM has both advantages and disadvantages relative to TDM. The main advantage is
that unlike TDM, FDM is not sensitive to propagation delays. Channel equalization
techniques needed for FDM systems are therefore not as complex as those for TDM
systems. Disadvantages of FDM include the need for bandpass filters, which are
relatively expensive and complicated to construct and design (remember that these filters
are usually used in the transmitters as well as the receivers). TDM, on the other hand,
uses relatively simple and less costly digital logic circuits. Another disadvantage of FDM
is that in many practical communication systems, the power amplifier in the transmitter
has nonlinear characteristics (linear amplifiers are more complex to build), and nonlinear
amplification leads to the creation of out-of-band spectral components that may interfere
with other FDM channels. Thus, it is necessary to use more complex linear amplifiers in
FDM systems.

Example 7.4—FDM for commercial FM radio

The frequency band from 88 MHz to 108 MHz is reserved over the public airwaves for
commercial FM broadcasting. The 88–108 MHz frequency band is divided into 200 kHz
sub-bands. As we saw in Chapter 6, the 200 kHz bandwidth of each sub-band is sufficient
for high-quality FM broadcast of music. The stations are identified by the center
frequency within their channel (e.g., 91.5 MHz, 103.7 MHz). This system can provide
radio listeners with their choice of up to 100 different radio stations.

FDM achieves the combining of several digital signals into one medium by sending
signals in several distinct frequency ranges over that medium. One of FDM's most
common applications is cable television. Only one cable reaches a customer's home but
the service provider can send multiple television channels or signals simultaneously over
that cable to all subscribers. Receivers must tune to the appropriate frequency (channel)
to access the desired signal.

Broadcast radio and television channels are separated in the frequency spectrum using
FDM. Each individual channel occupies a finite frequency range

Time-division multiplexing (TDM)

TDM involves sequencing groups of a few bits or bytes from each individual input
stream, one after the other, and in such a way that they can be associated with the
appropriate receiver. Time-division multiplexing (TDM) is a type of digital or (rarely)
analog multiplexing in which two or more signals or bit streams are transferred as sub-
channels in one communication channel, which are physically taking turns on the
channel. The time domain is divided into several recurrent timeslots of fixed length, one
for each sub-channel. A sample byte or data block of sub-channel 1 is transmitted during
timeslot 1, sub-channel 2 during timeslot 2, etc. One TDM frame consists of one timeslot
per sub-channel. After the last sub-channel the cycle starts all over again with a new
frame, starting with the second sample, byte or data block from sub-channel 1, etc.

Time Division Multiplexing is a technique where a short time sample of each

channel is inserted into the multiplexed data stream. Each channel is sampled in
turn, and then the sequence is repeated. The sample period has to be fast
enough to sample each channel according to the Nyquist Theory (2x highest
frequency), and to be able to sample all the other channels within that same time
period. It can be thought of as a very fast mechanical switch: selecting each
channel for a very short time, then going on to the next channel.
It's often practical to combine a set of low-bit-rate streams, each with a fixed and pre-
defined bit rate, into a single high-speed bit stream that can be transmitted over a single
channel. This technique is called time division multiplexing (TDM). It would be very
expensive if each low-bit-rate stream were assigned a costly physical channel (say, an
entire fiber optic line) that extended over a long distance.

Benefits of TDM

TDM is all about cost: fewer wires and simpler receivers are used to transmit data from
multiple sources to multiple destinations. TDM also uses less bandwidth than Frequency-
Division Multiplexing (FDM) signals, unless the bitrate is increased, which will
subsequently increase the necessary bandwidth of the transmission

Consider, for instance, a channel capable of transmitting 192 kbit/sec from Chicago to
New York. Suppose that three sources, all located in Chicago, each have 64 kbit/sec of
data that they want to transmit to individual users in New York. As shown in Figure 7-2,
the high-bit-rate channel can be divided into a series of time slots, and the time slots can
be alternately used by the three sources. The three sources are thus capable of
transmitting all of their data across the single, shared channel. Clearly, at the other end of
the channel (in this case, in New York), the process must be reversed (i.e., the system
must divide the 192 kbit/sec multiplexed data stream back into the original three 64
kbit/sec data streams, which are then provided to three different users). This reverse
process is called demultiplexing.

Figure 7-2—Time division multiplexing.

Choosing the proper size for the time slots involves a trade-off between efficiency and
delay. If the time slots are too small (say, one bit long) then the multiplexer must be fast
enough and powerful enough to be constantly switching between sources (and the
demultiplexer must be fast enough and powerful enough to be constantly switching
between users). If the time slots are larger than one bit, data from each source must be
stored (buffered) while other sources are using the channel. This storage will produce
delay. If the time slots are too large, then a significant delay will be introduced between
each source and its user. Some applications, such as teleconferencing and
videoconferencing, cannot tolerate long delays.

As shown in Example 7-2, the sources that are multiplexed may have different bit rates.
When this occurs, each source is assigned a number of time slots in proportion to its
transmission rate.

Example 7.1—The T1 system for wireline telephone networks

The T1 system is used for wireline long-distance service in North America and is an
excellent example of TDM. Speech from a telephone conversation is sampled once every
125 msec and each sample is converted into eight bits of digital data (see Chapter 8 for
more details). Using this technique, a transmission speed of 64,000 bits/sec is required to
transmit the speech. A T1 line is essentially a channel capable of transmitting at a speed
of 1.544 Mbit/sec. This is a much higher transmission speed than a single telephone
conversation needs, so TDM is used to allow a single T1 line to carry 24 different speech
signals between, say, two different telephone substations (called central offices) within a
city. As shown in Figure 7-3, the 1.544 Mbit/sec bit stream is divided into 193-bit frames.
The duration of each frame is

corresponding to the period between samples of the speech. Each frame is divided into 24
slots, which are each eight bits wide (corresponding to the number of bits needed to
digitize a speech sample). One additional bit at the end of the frame is used for signaling.
The eight bits of data corresponding to a sample of the speech are placed into one of the
24 slots in the frame.

For longer distances (say, between two large cities) higher-capacity channels are used
and multiple T1 lines are time division multiplexed onto the new channels. A T3 channel
for example, has a transmission speed of 44.736 Mbit/sec and uses TDM to carry 28 T1
lines (a total of 672 different speech signals) plus signaling. For more information on this
hierarchical multiplexing system, see BeIlamy [7.1].
 Figure 7-3—Time division multiplexing on a T1 line.

Example 7.2—TDM with sources having different data rates

Consider the case of three streams with bit rates of 8 kbit/sec,16 kbit/sec, and 24 kbit/sec,
respectively. We want to combine these streams into a single high-speed stream using
TDM. The high-speed stream in this case must have a transmission rate of 48 kbit/sec,
which is the sum of the bit rates of the three sources. To determine the number of time
slots to be assigned to each source in the multiplexing process. we must reduce the ratio
of the rates, 8:16:24, to the lowest possible form, which in this case is 1:2:3. The sum of
the reduced ratio is 6, which will then represent the minimum length of the repetitive
cycle of slot assignments in the multiplexing process. The solution is now readily
obtained: In each cycle of six time slots we assign one slot to Source A (8 kbit/sec), two
slots to Source B (16 kbit/sec), and three slots to Source: C (24 kbit/sec). Figure 7-4
illustrates this assignment, using “a” to indicate data from Source A, “b” to indicate data
from Source B, and “c” to indicate data from Source C.

Figure 7-4—Multiplexing input lines with different transmission speeds.

Example 7.3—A more complex TDM system

Consider a system with four low-bit-rate sources of 10 kbit/sec, 15 kbit/sec, 20 kbit/sec,

and 30 kbit/sec. Determine the slot assignments when the data streams are combined
using TDM.

Solution
The rate ratio 10:15:20:30 reduces to 2:3:4:6. The length of the cycle is therefore 2 + 3 +
4 + 6 = 15 slots. Within each cycle of 15 slots, we assigrn two slots to the 10 kbit/sec
source, three slots to the 15 kbit/sec source, four slots to the 20 kbit/sec source, and six
slots to the 30 kbit/sec source.

So far we have considered a form of TDM that is based on fixed slot assignments to each
of the low-bit-rate data streams. In other words, each stream has predefined slot positions
in the combined stream, and the receiver must be aware which slots belong to which
input stream. Both transmission ends, the transmitter and the receiver, must be perfectly
synchronized to the slot period. For this reason, the technique is usually called
synchronous TDM.

Synchronous TDM

Synchronous TDM is a system where the transmitter and the receiver both know exactly
which signal is being sent. Consider the following diagram:

Signal A ---> |---| |A|B|C|A|B|C| |------| ---> Signal A

Signal B ---> |TDM| --------------> |De-TDM| ---> Signal B
Signal C ---> |---| |------| ---> Signal C

In this system, starting at time-slice 0, every third time-slice is reserved for Signal A;
starting at time-slice 1, every third time-slice is reserved for Signal B; and starting at
time-slice 2, every third time-slice is reserved for Signal C. In this situation, the receiver
(De-TDM) needs only to switch after the signal on each time-slice is received.

The data flow of each input connection is divided into units where each input occupies
one input time slot. Each input connection has a time slot alloted in the output
irrespective of the fact whether it is sending data or not.

A -----|A3|A2|A1|---> |---| .............|C3|B3|A3|

C2|B2|A2|C1|B1|A1| |------| ---> A
| | | | | |
|
B -----|B3|B2|B1|---> |MUX|
-------------|--------|--------|----------> |De-MUX| ---> B
| | | | | |
|
C -----|C3|C2|C1|---> |---| | |
| |------| ---> C
<-->
<-------->
Bit Interval Frame
(x seconds)
Sync TDM is inefficient when one or more input lines have no data to send. Thus, it is
used with lines with high data traffic.

Sampling rate is same for all signals. Maximum sampling rate = twice the maximum
frequency all the signals

There is another important version of TDM, usually referred to as statistical TDM.

Statistical TDM is useful for applications in which the low-bit-rate streams have speeds
that vary in time. For example, a low-bit-rate stream to a single terminal in a computer
network may fluctuate between 2 kbit/sec and 50 kbit/sec during an active connection
session (we've all seen variable speeds during Internet connections, for instance). If we
assign the stream enough slots for its peak rate (that is, for 50 kbit/sec), then we will be
wasting slots when the rate drops well below the peak value. This waste can be especially
significant if the system has many variable-speed low-bit-rate streams.

Statistical TDM works by calculating the average transmission rates of the streams to be
combined, and then uses a high-speed multiplexing link with a transmission rate that is
equal to (or slightly greater than) the statistical average of the combined streams. Since
the transmission rates from each source are variable, we no longer assign a fixed number
of time slots to each data stream. Rather, we dynamically assign the appropriate number
of slots to accommodate the current transmission rates from each stream. Because the
combined rate of all the streams will also fluctuate in time between two extreme values,
we need to buffer the output of the low-bit-rate streams when the combined rate exceeds
the transmission rate of the high-speed link.

With statistical TDM, we are no longer relying on synchronized time slots with fixed
assignments for each input stream, as we did with synchronous TDM. So how does the
demultiplexer in statistical TDM know which of the received bits belongs to which data
stream? Prior to transmission, we divide each stream of bits coming from a source into
fixed-size blocks. We then add a small group of bits called a header to each block, with
the header containing the addresses of the source and intended user for that block. The
block and the header are then transmitted together across the channel. Combined, the
block and header are called a packet.

Actually, the header may contain other information besides the source and user addresses,
such as extra bits for error control (see Chapter 10) or additional bits for link control
(used, for example, to indicate the position of a particular block in a sequence of blocks
coming from the same user, or to indicate priority level for a particular message). Extra
bits can also be added to the beginning and end of a block for synchronization; a
particular pattern of bits, called a start flag, can be used in the header to mark the start of
a block, and another particular pattern of bits, called an end flag, can be used to conclude
the block. Each block transmitted across the channel thus contains a group of information
bits that the user wants, plus additional bits needed by the system to ensure proper
transmission. These additional bits, while necessary to system operation, reduce the
effective transmission rate on the channel. Figures 7-5 and 7-6 present the statistical
TDM technique and the structure of a typical packet.

Figure 7-5—Statistical TDM.

Figure 7-6—Structure of a typical statistical TDM packet

Statistical Time Division Multiplexing uses intelligent devices that are capable of
identifying when a terminal is idle. They allocate time only to lines when required. This
means that more lines can be connected to a transmission medium because this device
statistically compensates for normal idle time (in data communication lines). Newer
STDM units provide additional capabilities: data compression, line priority, mixed speed
lines, host port sharing, network port control, automatic speed detection and much more.

Statistical TDM

Synchronous TDM is beneficial because the receiver and transmitter can both cost very
little. However, consider the most well-known network: the Internet. In the Internet, a
given computer might have a data rate of 1kbps when hardly anything is happening, but
might have a data rate of 100kbps when downloading a large file from a fast server. How
are the time-slices divided in this instance? If every time slice is made big enough to hold
100Kbps, when the computer isn't downloading any data, all of that time and electricity
will be wasted. If every time-slice is only big enough for the minimum case, the time
required to download bigger files will be greatly increased.

The solution to this problem is called Statistical TDM, and is the solution that the
Internet currently uses. In Statistical TDM, each data item, known as the payload (we
used time-slices to describe these earlier), is appended with a certain amount of
information about who sent it, and who is supposed to receive it (the header). The
combination of a payload and a header is called a packet. Packets are like envelopes in
the traditional "snail mail" system: Each packet contains a destination address and a
return address as well as some enclosed data. Because of this, we know where each
packet was sent from and where it is going.

The downside to statistical TDM is that the sender needs to be smart enough to write a
header, and the receiver needs to be smart enough to read the header and (if the packet is
to be forwarded,) send the packet toward its destination.

Link Utilization

Statistical multiplexing attempts to maximize the use of a communication path. The study
of this is often called queuing theory. A queue is simply a line of customers or packets
waiting to be served. Under most circumstances, the arrival rate is unpredictable and
therefor follows a random or Poisson distribution pattern, whereas the service time is
constant.

The utilization or fraction of time actually used by a packet multiplexing system to

process packets is given by:
The queue length or average number of items waiting to be served is given by:

Example
A T1 link has been divided into a number of 9.6 Kbps channels and has a
combined user data rate of 1.152 Mbps. Access to this channel is offered to 100
customers, each requiring 9.6 Kbps data 20% of the time. If the user arrival time
is strictly random find the T1 link utilization.
Solution
The utilization or fraction of time used by the system to process packets is given
by:

A 24 channel system dedicated to DATA, can place five 9.6 Kbps customers in
each of 23 channels, for a total of 115 customers. In the above statistical link, 100
customers created an average utilization of 0.167 and were easily fitted, with
room to spare if they transmit on the average 20% of the time. If however, the
 customer usage were not randomly distributed, then the above analysis would
have to be modified.

This example shows the potential for statistical multiplexing. If channels were assigned
on a demand basis (only when the customer had something to send), a single T1 may be
able to support hundreds of low volume users.

A utilization above 0.8 is undesirable in a statistical system, since the slightest variation
in customer requests for service would lead to buffer overflow. Service providers
carefully monitor delay and utilization and assign customers to maximize utilization and
minimize cost.

Packets

Packets will be discussed in greater detail once we start talking about digital networks
(specifically the Internet). Packet headers not only contain address information, but may
also include a number of different fields that will display information about the packet.
Many headers contain error-checking information (checksum, Cyclic Redundancy
Check) that enables the receiver to check if the packet has had any errors due to
interference, such as electrical noise.

Duty Cycles

Duty cycle is defined as " the time that is effectively used to send or receive the data,
expressed as a percentage of total period of time." The more the duty cycle , the more
effective transmission or reception.

We can define the pulse width, τ, as being the time that a bit occupies from within it's
total alloted bit-time Tb. If we have a duty cycle of D, we can define the pulse width as:

τ = DTb

Where:

The pulse width is equal to the bit time if we are using a 100% duty cycle

code division multiplexing (CDM).

Code division multiplexing (CDM) is a technique in which each channel transmits its bits
as a coded channel-specific sequence of pulses. This coded transmission typically is
accomplished by transmitting a unique time-dependent series of short pulses, which are
placed within chip times within the larger bit time. All channels, each with a different
code, can be transmitted on the same fiber and asynchronously demultiplexed

Multiple access methods

Generally, a fixed amount of spectrum is allocated for a cellular system by a national

regulator (For example, the Federal Communications Commission in the United States).
Thus, Multiple Access techniques are necessary in order to allow many users to share the
available spectrum in an efficient way.

Multiple Access techniques specify the way signals from different sources are to be
combined efficiently for transmission over a given radio frequency band and then
separated at the destination without mutual interference.

a channel access method or multiple access method allows several terminals connected
to the same multi-point transmission medium to transmit over it and to share its capacity.
A channel-access scheme is based on a multiplexing method, that allows several data
streams or signals to share the same communication channel or physical medium.

These are the fundamental forms of channel access schemes:

• The frequency division multiple access (FDMA) channel-access scheme is based

on the frequency-division multiplex (FDM) scheme, which provides different
frequency bands to different data-streams - in the FDMA case to different users or
nodes. An example of FDMA systems were the first-generation (1G) cell-phone
systems. A related technique is wave-length division multiple access (WDMA),
based on wavelength division multiplex (WDM), where different users get
different colors in fiber-optical communication.
• The time division multiple access (TDMA) channel access scheme is based on the
time division multiplex (TDM) scheme, which provides different time-slots to
different data-streams (in the TDMA case to different transmitters) in a cyclically
repetitive frame structure. For example, user 1 may use time slot 1, user 2 time
slot 2, etc until the last user. Then it starts all over again.
• The code division multiple access (CDMA) scheme is based on spread spectrum.
An example is the 3G cell phone system.
• Space division multiple access (SDMA).
• Packet mode multiple-access is typically also based on time-domain multiplexing,
but not in a cyclically repetitive frame structure, and therefore it is not considered
as TDM or TDMA. Due to its random character it can be categorised as statistical
multiplexing methods, making it possible to provide dynamic bandwidth
allocation.

Time Division multiple Access

How TDMA works

Time division multiple access takes advantage of the digitization of the signals in order to
accommodate information from several users within one frequency channel.

Nyquist's sampling theorem assures that if a band limited signal with bandwidth W is
sampled at rate of at least Ws=2*W then the signal can be fully reconstructed from its
samples and no information is lost. Thus, the signal's samples can be transmitted instead
of the signal itself. But now, the time between transmitted samples can be utilized to
transmit samples of other signals in order to increase the capacity of the frequency
channel. This is a simplified conceptual description of how TDMA works.

More elaborately, when a user wishes to transmit an analog signal (voice), the signal is
sampled, quantized and digitized in a process that is called PCM-Pulse Code Modulation
(If the signal is already digitized (data signal) this is unnecessary). As a result the signal
is converted into a stream of digital information. The stream is compressed by a digital
speech code into bursts 1/n of their original length. The burst takes only 1/n of the airtime
required to transmit the original audio signal, leaving n-1/n of the time for the other u.
The digital burst is then modulated into the channel's frequency and in the time slot that
was allocated for the user the burst is transmitted. The channel's frequency and the
allocated time slot for the user are informed to the mobile user by the base station when
the call is set up via a control channel.

TDMA is a store and burst system. Incoming user traffic is stored in memory and when a
user's turn comes up, this accumulated traffic is transmitted in a digital burst.
In TDMA, the transmission is divided to frames, which contain several time slots. In each
time slot, a different user transmits his digital burst. This method of multiplexing that
combines data streams by assigning each stream a different time slot in a set is called
Time Division Multiplexing (TDM) (this technique is also used in T1/E1 channels).

The number of time slots in a frame (n) is standard dependent. Effectively, TDMA
implementations that use n:1 multiplexing (i.e. divide the channel's given bandwidth into
n time slots) increase capacity by n. North American cellular standards IS-54 and IS-136,
for example, triple the capacity of cellular frequencies by dividing a 30-kHz channel into
three time slots, enabling three different users to occupy it at the same time.
Currently, systems are in place that allows six times capacity. In the future, with the
utilization of hierarchical cells, intelligent antennas, and adaptive channel allocation, the
capacity should approach 40 times analog capacity.

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Enhanced TDMA

TDMA substantially improved upon the efficiency of analog cellular. However, like
FDMA, it had the weakness that it wasted bandwidth: the time slot was allocated to a
specific conversation whether or not anyone was speaking at that moment.

An enhanced version of TDMA (ETDMA) improve the bandwidth utilization by

assigning time slots to users dynamically instead of waiting to determine whether a user
is transmitting. ETDMA sends data through those pauses which normal speech contains.
When a user has something to transmit, he puts one bit in a buffer queue. The system
scans the buffer, notices that the user has something to transmit, and allocates bandwidth
accordingly. If a user has nothing to transmit, he is skipped. So, instead of being
arbitrarily assigned, time is allocated according to need. If partners in a phone
conversation do not speak over one another, this technique can almost double the
bandwidth utilization efficiency of TDMA, making it almost 10 times as efficient as
analog transmission.

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TDMA Advantages and Disadvantages

Advantages

Personal Communication

Services such as SMS - short message data, fax, voice band data, and also
multimedia, video-conferencing, which is bandwidth-intensive application. All of
these can be supplied by the TDMA because of the ability to carry data rates of 64
Kbps to 120 Mbps.

Efficiency

This technology increases the efficiency of transmission.

Interference

In this technology, the users will not experience interference from other
simultaneous transmissions because of the separation in time between different
users.

Battery life

Because the mobile is only transmitting a portion of the time, this extends the
battery life and as a result of that the talk time.

Cost

While upgrading a current analog system to digital, using TDMA is the advisable
technology for that as the most cost-effective.

Installation

This technology installation offers substantial savings in base-station space,

equipment and maintenance.

Utilization of HCS

TDMA supply an efficient utilization of HCS - Hierarchical Cell Structures by

offering pico, micro and macrocells.

Service Compatibility

This technology allows Service Compatibility with the use of dual-mode

handsets. This can be done because of the TDMA's inherit compatibility with
FDMA analog system.
 Disadvantages

Predefined Time Slot

Each user has a predefined time slot, but the users are not allocated a time slot
while they are roaming from one cell to another - this might cause a call to be
disconnected in case that all time slots in the next cell are already occupied.

Another problem with predefined timeslots is that a fixed and predefined number
of users will have channel access. Thus, if all time slots are already occupied, new
users wishing to transmit and get access rights won't be able to do so (their call
will be disconnected).

Multipath Distortion

This technology is subject to multipath distortion - An Interference might be

caused in case that a signal which is coming from a tower to a handset might also
come from any one of several directions and will be bounced off several different
buildings before arriving to it's target.

Time division multiple access (TDMA) is a channel access method for shared medium
networks. It allows several users to share the same frequency channel by dividing the
signal into different time slots. The users transmit in rapid succession, one after the other,
each using his own time slot. This allows multiple stations to share the same transmission
medium (e.g. radio frequency channel) while using only a part of its channel capacity.
TDMA is a type of Time-division multiplexing, with the special point that instead of
having one transmitter connected to one receiver, there are multiple transmitters.
1. The Digital Advantage
All multiple access techniques depend on the adoption of digital technology.
Digital technology is now the standard for the public telephone system where all
analog calls are converted to digital form for transmission over the backbone.
Digital has a number of advantages over analog transmission:

• It economizes on bandwidth.
• It allows easy integration with personal communication systems (PCS)
devices.
• It maintains superior quality of voice transmission over long distances.
• It is difficult to decode.
• It can use lower average transmitter power.
 • It enables smaller and less expensive individual receivers and
transmitters.
• It offers voice privacy.

Equalisation

Equalization, equalisation or EQ is the process of using passive or active electronic

elements or digital algorithms for the purpose of altering (originally flattening) the
frequency response characteristics of a system.[1]

The term generally implies Amplitude equalization, but any frequency dependent
response characteristic could be equalized. There are many kinds of EQ. Each has a
different pattern of attenuation or boost. A peaking equalizer raises or lowers a range of
frequencies around a central point in a bell shape.

Guard interval

In telecommunications, guard intervals are used to ensure that distinct transmissions do

not interfere with one another. These transmissions may belong to different users (as in
TDMA) or to the same user (as in OFDM). The purpose of the guard interval is to
introduce immunity to propagation delays, echoes and reflections, to which digital data is
normally very sensitive. In TDMA, each user's timeslot ends with a guard period, to
avoid data loss and to reduce interference to the following user, caused by propagation
delay. Thus a user's timeslot is protected from interference from the preceding user, by
the guard interval (guard period) at the end of that preceding user's timeslot. It is a
common misconception that each TDMA timeslot begins with a guard interval, however
the ITU Technical Specifications (such as GSM 05.05) clearly define the guard period as
being at the end of each timeslot, thus providing protection against data loss within that
timeslot, and protection against interference to the following timeslot.

In the GSM system, the synchronization of the mobile phones is achieved by sending
timing advance commands from the base station which instructs the mobile phone to
transmit earlier and by how much. This compensates for the propagation delay resulting
from the light speed velocity of radio waves. The mobile phone is not allowed to transmit
for its entire time slot, but there is a guard interval at the end of each time slot. As the
transmission moves into the guard period, the mobile network adjusts the timing advance
to synchronize the transmission.

Initial synchronization of a phone requires even more care. Before a mobile transmits
there is no way to actually know the offset required. For this reason, an entire time slot
has to be dedicated to mobiles attempting to contact the network (known as the RACH in
GSM). The mobile attempts to broadcast at the beginning of the time slot, as received
from the network. If the mobile is located next to the base station, there will be no time
delay and this will succeed. If, however, the mobile phone is at just less than 35 km from
the base station, the time delay will mean the mobile's broadcast arrives at the very end of
the time slot. In that case, the mobile will be instructed to broadcast its messages starting
nearly a whole time slot earlier than would be expected otherwise. Finally, if the mobile
is beyond the 35 km cell range in GSM, then the RACH will arrive in a neighboring time
slot and be ignored. It is this feature, rather than limitations of power, that limits the
range of a GSM cell to 35 km when no special extension techniques are used. By
changing the synchronization between the uplink and downlink at the base station,
however, this limitation can be overcome.

Random Access Channel (RACH) is used in mobile phones or other wireless device on
a TDMA-based network when it needs to get the attention of a base station in order to
initially synchronize its transmission with the base station. Random Access Channel is a
shared channel that is used by Wireless Access terminals to Access the Access
Network(TDMA/FDMA,and CDMA based network) especially for initial Access and
Bursty data transmission. A key feature of a Random Access Channel is that messages
are not scheduled (compared to for example a "Dedicated Channel" in UMTS, that is
assigned exclusively to one user at a time). There is no certainty, that only a single device
makes a connection attempt at one time, and collisions can result.

The Advantages of TDMA

In addition to increasing the efficiency of transmission, TDMA offers a number of
other advantages over standard cellular technologies. First and foremost, it can
be easily adapted to the transmission of data as well as voice communication.
TDMA offers the ability to carry data rates of 64 kbps to 120 Mbps (expandable in
multiples of 64 kbps). This enables operators to offer personal communication-
like services including fax, voiceband data, and short message services (SMSs)
as well as bandwidth-intensive applications such as multimedia and
videoconferencing.

Unlike spread-spectrum techniques which can suffer from interference among

the users all of whom are on the same frequency band and transmitting at the
same time, TDMA’s technology, which separates users in time, ensures that they
will not experience interference from other simultaneous transmissions.

TDMA also provides the user with extended battery life and talk time since the
mobile is only transmitting a portion of the time (from 1/3 to 1/10) of the time
during conversations.

TDMA installations offer substantial savings in base-station equipment, space,

and maintenance, an important factor as cell sizes grow ever smaller.

TDMA is the most cost-effective technology for upgrading a current analog

system to digital.
TDMA is the only technology that offers an efficient utilization of hierarchical cell
structures (HCSs) offering pico, micro, and macrocells. HCSs allow coverage for
the system to be tailored to support specific traffic and service needs. By using
this approach, system capacities of more than 40-times AMPS can be achieved
in a cost-efficient way.

Because of its inherent compatibility with FDMA analog systems, TDMA allows
service compatibility with the use of dual-mode handsets.

The Disadvantages of TDMA

One of the disadvantages of TDMA is that each user has a predefined time slot.
However, users roaming from one cell to another are not allotted a time slot.
Thus, if all the time slots in the next cell are already occupied, a call might well be
disconnected. Likewise, if all the time slots in the cell in which a user happens to
be in are already occupied, a user will not receive a dial tone.

Another problem with TDMA is that it is subjected to multipath distortion. A signal

coming from a tower to a handset might come from any one of several directions.
It might have bounced off several different buildings before arriving.

Dynamic TDMA

In dynamic time division multiple access, a scheduling algorithm dynamically reserves

a variable number of time slots in each frame to variable bit-rate data streams, based on
the traffic demand of each data stream.

The main difference between tdm and tdma (also fdm/fdma, etc) is that with
tdm (also fdm, etc.) the signals multiplexed (i.e. sharing a resource) come from
the same node, whereas for tdma (also fdm, etc.) the signals multiplexed come
from different sources/transmitters. - Time Division Multiplexing (TDM) imply
partitioning the bandwidth of the channel connecting two nodes into finite set of
time slots. - Time Division multiple Access (TDMA) imply partitioning the
bandwidth of a channel shared by many nodes, typically an infrastructure node
and several mobile nodes, where each node gets to access its dedicated time
slot.

The main difference between tdm and tdma (also fdm/fdma,

etc) is that
with tdm (also fdm, etc.) the signals multiplexed (i.e.
sharing a
resource) come from the same node, whereas for tdma (also
fdm, etc.) the
signals multiplexed come from different
sources/transmitters. This is
the reason why we use tdm for downlink (forward link) and
tdma for

uplink (reverse link).

Frequency division multiple access

FDMA divides the given spectrum into channels by the frequency domain. Each
phone call is allocated one channel for the entire duration of the call. In the figure
above, each band represents one call.

Signalling